From 67cabe6c8ba05b452ee5ec31caa2ed262f19e639 Mon Sep 17 00:00:00 2001
From: zhifu gao <zhifu.gzf@alibaba-inc.com>
Date: 星期四, 27 四月 2023 17:30:58 +0800
Subject: [PATCH] Merge pull request #434 from alibaba-damo-academy/dev_websocket

---
 funasr/runtime/python/websocket/README.md |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/funasr/runtime/python/websocket/README.md b/funasr/runtime/python/websocket/README.md
index 723782f..c5284ff 100644
--- a/funasr/runtime/python/websocket/README.md
+++ b/funasr/runtime/python/websocket/README.md
@@ -1,4 +1,4 @@
-# Using funasr with websocket
+# Service with websocket-python
 We can send streaming audio data to server in real-time with grpc client every 300 ms e.g., and get transcribed text when stop speaking.
 The audio data is in streaming, the asr inference process is in offline.
 

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